Chapter 14 VoIP
102
PRACK (RFC 3262)
RFC 3262 defines a mechanism to provide reliable transmission of SIP provisional
response messages, which convey information on the processing progress of the
request. This uses the option tag 100rel and the Provisional Response
ACKnowledgement (PRACK) method.
Select
Supported
or
Required
to have the Router include a SIP Require/Supported
header field with the option tag 100rel in all INVITE requests. When the Router
receives a SIP response message indicating that the phone it called is ringing, the
Router sends a PRACK message to have both sides confirm the message is received.
If you select
Supported
, the peer device supports the option tag 100rel to send
provisional responses reliably.
If you select
Required
, the peer device requires the option tag 100rel to send
provisional responses reliably.
Select
Disabled
to turn off this function.
DNS SRV Enabled
(RFC 3263)
Select this to have the Router query your ISP’s DNS server for a list of any available SIP
servers that it maintains. This is useful if your static SIP server experiences difficulties,
making it hard for your IP phone users to make SIP calls.
Session Timer
(RFC 4028)
Select this to have the Router support RFC 4028.
This makes sure that SIP sessions do not hang and the SIP line can always be available
for use.
RTP Port Range
Enter the listening port number(s) for RTP traffic provided by your VoIP service
provider.
Start Port
End Port
If your VoIP service provider gave you this information. Otherwise, keep the default
values.
To enter one port number, enter the port number in the
Start Port
and
End Port
fields.
To enter a range of ports,
• enter the port number at the beginning of the range in the
Start Port
field.
• enter the port number at the end of the range in the
End Port
field.
DTMF Mode
Control how the Router handles the tones that your telephone makes when you push
its buttons. You should use the same mode your VoIP service provider uses.
RFC2833
- send the DTMF tones in RTP packets.
Inband
- send the DTMF tones in the voice data stream. This method works best when
you are using a codec that does not use compression (like G.711). Codecs that use
compression (like G.726) can distort the tones.
SIPInfo
- send the DTMF tones in SIP messages.
Transport Type
The transport layer protocol used for SIP is
UDP
.
FAX Option
This field controls how the Router handles fax messages.
G711 Fax
Passthrough
Select this if the Router should use G.711 to send fax messages. The peer devices
must also use G.711.
Table 61
SIP > SIP Service Provider: Edit (continued)
LABEL
DESCRIPTION
